1. Field of the Invention
The present invention relates to a method of controlling a transmission over a local area network, and more specifically, to a method of using an RTCP/TCP/IP protocol to transmit QoS packets during multimedia data transmission.
2. Description of the Prior Art
With the rapid development of computer technology, costs are reduced and functions improved; as a result, many families now own two more computers. Through a simple local area network established by two computers at home, a user at home is able to share a printer or an optical drive, play on-line games, and even share programs. For example, some programs can be installed in one computer that has more storage capacity or higher speed CPU, and can be shared by the other one via the local area network, instead of being installed on both computers. Therefore, a family only has to own a single large-capacity computer for storing multimedia files of several gigabytes, and share such files through the local area network.
Please refer to FIG. 1, showing a schematic diagram of a conventional local area network 10. The local area network 10 comprises a plurality of hosts, one as sending end 12, and the others as receiving ends 14.
Suppose that an on-line video home meeting is held via the local area network 10. Generally speaking, because a higher transmission quality is required when real-time audio and video data packets are transmitted between the sending end 12 and the receiving ends 14, an RTP/UDP/IP protocol is used to send audio and video data packets. The UDP/IP protocol is a simple but unreliable data packet transmission protocol; “unreliable” in the sense that the data packets from the sending end 12 are not guaranteed to either reach the receiving end 14 or arrive in a correct sequence. Fortunately, for a small local area network, the arrival ratio of the data packets using the UDP/IP protocol is close to 100%. Therefore, the UDP/IP protocol, which has shorter packets than TCP/IP protocol packets due to less and simpler parameter settings, is commonly used in multimedia data transmission. Additionally, the sending end 12 can multicast data packets to given ports of the receiving ends 14 on the identical local network 10. Furthermore, a Real-Time Transport Protocol (RTP) in cooperation with UDP/IP is used for packaging the audio and video data. An RTP header within the audio and video data packets provides essential timing information and an order number so that the receiving end 14 recognizes whether a data packet is lost or that the data packets have arrived in order during transmission so that the receiving ends 14 (i.e. all the meeting participants) can exactly recombine the received packets and estimate the number of lost packets. Therefore, the RTP header further contains some information for a data-compression protocol, such as PCM and ADPCM. In this way, if a huge amount of video and audio is transmitted in the data stream, RTP/UDP/IP is usually used for transmitting the data stream. The basic principle of RTP, defined in RFC 1889, is well known in the art.
Before sending data packets, the sending end 12 has to get a multicast address and two ports, one for RTP packets (video packet) and the other one for RTCP (RTP control protocol) packets, i.e. Quality of Service (QoS) control packets. The purpose of transmitting QoS packets is to ensure the transmission quality will not be limited by bandwidth, delay, jitter, and packet loss. Generally speaking, the sending end 12 and the receiving end 14 send QoS packets using RTCP/UDP/IP there and between in a regular period. In addition, RTCP is used for synchronization of audio and video packets.
However, generally speaking, no router or only one or two routers are arranged in the conventional home local area network 10 for peer-to-peer data transmission. Under the UDP/IP circumstance, such parameters as “jitter” and “round-trip-time” within the RTCP header of the QoS packet are useless, where “jitter” indicates an arrived time difference between each packet, and “round-trip-time” indicates a back and forth interval of each packet. That is because although each router can analyze the UDP/IP header within the QoS packet to maintain the data flow control among the routers, few routers are present in the conventional home local area network 10, meaning that the QoS packet will pass through one or two routers or no router at all.
As mentioned above, using UDP/IP to transmit the packets is not guaranteed to arrive at the destination or arrive in a correct order. When several users simultaneously share the bandwidth of the local network 10, the transmitted packets will probably be lost or delayed due to network congestion. In addition, for UDP/IP, packets that do not arrive at their destination are not resent so that the loss of a transmitted QoS packet using UDP/IP is possible. Even if the QoS packet arrives at its destination, “jitter” and “round-trip-time” parameters are still ineffective because the QoS packet using UDP/IP does not pass through only one or two routers or no router at all, resulting in invalid “jitter” and “round-trip-time” parameters. Also, as well, a local network 10 using UDP/IP fails to adjust the audio and video packets of transmission quality between the sending end 12 and any receiving end 14, resulting in bad playing quality of the audio and video packets.